rtmp: do not hardcode invoke numbers
[libav.git] / libavformat / rtmpproto.c
1 /*
2 * RTMP network protocol
3 * Copyright (c) 2009 Kostya Shishkov
4 *
5 * This file is part of Libav.
6 *
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file
24 * RTMP protocol
25 */
26
27 #include "libavcodec/bytestream.h"
28 #include "libavutil/avstring.h"
29 #include "libavutil/intfloat_readwrite.h"
30 #include "libavutil/lfg.h"
31 #include "libavutil/sha.h"
32 #include "avformat.h"
33 #include "internal.h"
34
35 #include "network.h"
36
37 #include "flv.h"
38 #include "rtmp.h"
39 #include "rtmppkt.h"
40 #include "url.h"
41
42 //#define DEBUG
43
44 /** RTMP protocol handler state */
45 typedef enum {
46 STATE_START, ///< client has not done anything yet
47 STATE_HANDSHAKED, ///< client has performed handshake
48 STATE_RELEASING, ///< client releasing stream before publish it (for output)
49 STATE_FCPUBLISH, ///< client FCPublishing stream (for output)
50 STATE_CONNECTING, ///< client connected to server successfully
51 STATE_READY, ///< client has sent all needed commands and waits for server reply
52 STATE_PLAYING, ///< client has started receiving multimedia data from server
53 STATE_PUBLISHING, ///< client has started sending multimedia data to server (for output)
54 STATE_STOPPED, ///< the broadcast has been stopped
55 } ClientState;
56
57 /** protocol handler context */
58 typedef struct RTMPContext {
59 URLContext* stream; ///< TCP stream used in interactions with RTMP server
60 RTMPPacket prev_pkt[2][RTMP_CHANNELS]; ///< packet history used when reading and sending packets
61 int chunk_size; ///< size of the chunks RTMP packets are divided into
62 int is_input; ///< input/output flag
63 char playpath[256]; ///< path to filename to play (with possible "mp4:" prefix)
64 char app[128]; ///< application
65 ClientState state; ///< current state
66 int main_channel_id; ///< an additional channel ID which is used for some invocations
67 uint8_t* flv_data; ///< buffer with data for demuxer
68 int flv_size; ///< current buffer size
69 int flv_off; ///< number of bytes read from current buffer
70 RTMPPacket out_pkt; ///< rtmp packet, created from flv a/v or metadata (for output)
71 uint32_t client_report_size; ///< number of bytes after which client should report to server
72 uint32_t bytes_read; ///< number of bytes read from server
73 uint32_t last_bytes_read; ///< number of bytes read last reported to server
74 int skip_bytes; ///< number of bytes to skip from the input FLV stream in the next write call
75 uint8_t flv_header[11]; ///< partial incoming flv packet header
76 int flv_header_bytes; ///< number of initialized bytes in flv_header
77 int nb_invokes; ///< keeps track of invoke messages
78 } RTMPContext;
79
80 #define PLAYER_KEY_OPEN_PART_LEN 30 ///< length of partial key used for first client digest signing
81 /** Client key used for digest signing */
82 static const uint8_t rtmp_player_key[] = {
83 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
84 'F', 'l', 'a', 's', 'h', ' ', 'P', 'l', 'a', 'y', 'e', 'r', ' ', '0', '0', '1',
85
86 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
87 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
88 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
89 };
90
91 #define SERVER_KEY_OPEN_PART_LEN 36 ///< length of partial key used for first server digest signing
92 /** Key used for RTMP server digest signing */
93 static const uint8_t rtmp_server_key[] = {
94 'G', 'e', 'n', 'u', 'i', 'n', 'e', ' ', 'A', 'd', 'o', 'b', 'e', ' ',
95 'F', 'l', 'a', 's', 'h', ' ', 'M', 'e', 'd', 'i', 'a', ' ',
96 'S', 'e', 'r', 'v', 'e', 'r', ' ', '0', '0', '1',
97
98 0xF0, 0xEE, 0xC2, 0x4A, 0x80, 0x68, 0xBE, 0xE8, 0x2E, 0x00, 0xD0, 0xD1, 0x02,
99 0x9E, 0x7E, 0x57, 0x6E, 0xEC, 0x5D, 0x2D, 0x29, 0x80, 0x6F, 0xAB, 0x93, 0xB8,
100 0xE6, 0x36, 0xCF, 0xEB, 0x31, 0xAE
101 };
102
103 /**
104 * Generate 'connect' call and send it to the server.
105 */
106 static void gen_connect(URLContext *s, RTMPContext *rt, const char *proto,
107 const char *host, int port)
108 {
109 RTMPPacket pkt;
110 uint8_t ver[64], *p;
111 char tcurl[512];
112
113 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 4096);
114 p = pkt.data;
115
116 ff_url_join(tcurl, sizeof(tcurl), proto, NULL, host, port, "/%s", rt->app);
117 ff_amf_write_string(&p, "connect");
118 ff_amf_write_number(&p, 1.0);
119 ff_amf_write_object_start(&p);
120 ff_amf_write_field_name(&p, "app");
121 ff_amf_write_string(&p, rt->app);
122
123 if (rt->is_input) {
124 snprintf(ver, sizeof(ver), "%s %d,%d,%d,%d", RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1,
125 RTMP_CLIENT_VER2, RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
126 } else {
127 snprintf(ver, sizeof(ver), "FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
128 ff_amf_write_field_name(&p, "type");
129 ff_amf_write_string(&p, "nonprivate");
130 }
131 ff_amf_write_field_name(&p, "flashVer");
132 ff_amf_write_string(&p, ver);
133 ff_amf_write_field_name(&p, "tcUrl");
134 ff_amf_write_string(&p, tcurl);
135 if (rt->is_input) {
136 ff_amf_write_field_name(&p, "fpad");
137 ff_amf_write_bool(&p, 0);
138 ff_amf_write_field_name(&p, "capabilities");
139 ff_amf_write_number(&p, 15.0);
140 ff_amf_write_field_name(&p, "audioCodecs");
141 ff_amf_write_number(&p, 1639.0);
142 ff_amf_write_field_name(&p, "videoCodecs");
143 ff_amf_write_number(&p, 252.0);
144 ff_amf_write_field_name(&p, "videoFunction");
145 ff_amf_write_number(&p, 1.0);
146 }
147 ff_amf_write_object_end(&p);
148
149 pkt.data_size = p - pkt.data;
150
151 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
152 ff_rtmp_packet_destroy(&pkt);
153 }
154
155 /**
156 * Generate 'releaseStream' call and send it to the server. It should make
157 * the server release some channel for media streams.
158 */
159 static void gen_release_stream(URLContext *s, RTMPContext *rt)
160 {
161 RTMPPacket pkt;
162 uint8_t *p;
163
164 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
165 29 + strlen(rt->playpath));
166
167 av_log(s, AV_LOG_DEBUG, "Releasing stream...\n");
168 p = pkt.data;
169 ff_amf_write_string(&p, "releaseStream");
170 ff_amf_write_number(&p, ++rt->nb_invokes);
171 ff_amf_write_null(&p);
172 ff_amf_write_string(&p, rt->playpath);
173
174 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
175 ff_rtmp_packet_destroy(&pkt);
176 }
177
178 /**
179 * Generate 'FCPublish' call and send it to the server. It should make
180 * the server preapare for receiving media streams.
181 */
182 static void gen_fcpublish_stream(URLContext *s, RTMPContext *rt)
183 {
184 RTMPPacket pkt;
185 uint8_t *p;
186
187 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
188 25 + strlen(rt->playpath));
189
190 av_log(s, AV_LOG_DEBUG, "FCPublish stream...\n");
191 p = pkt.data;
192 ff_amf_write_string(&p, "FCPublish");
193 ff_amf_write_number(&p, ++rt->nb_invokes);
194 ff_amf_write_null(&p);
195 ff_amf_write_string(&p, rt->playpath);
196
197 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
198 ff_rtmp_packet_destroy(&pkt);
199 }
200
201 /**
202 * Generate 'FCUnpublish' call and send it to the server. It should make
203 * the server destroy stream.
204 */
205 static void gen_fcunpublish_stream(URLContext *s, RTMPContext *rt)
206 {
207 RTMPPacket pkt;
208 uint8_t *p;
209
210 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0,
211 27 + strlen(rt->playpath));
212
213 av_log(s, AV_LOG_DEBUG, "UnPublishing stream...\n");
214 p = pkt.data;
215 ff_amf_write_string(&p, "FCUnpublish");
216 ff_amf_write_number(&p, ++rt->nb_invokes);
217 ff_amf_write_null(&p);
218 ff_amf_write_string(&p, rt->playpath);
219
220 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
221 ff_rtmp_packet_destroy(&pkt);
222 }
223
224 /**
225 * Generate 'createStream' call and send it to the server. It should make
226 * the server allocate some channel for media streams.
227 */
228 static void gen_create_stream(URLContext *s, RTMPContext *rt)
229 {
230 RTMPPacket pkt;
231 uint8_t *p;
232
233 av_log(s, AV_LOG_DEBUG, "Creating stream...\n");
234 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 25);
235
236 p = pkt.data;
237 ff_amf_write_string(&p, "createStream");
238 ff_amf_write_number(&p, ++rt->nb_invokes);
239 ff_amf_write_null(&p);
240
241 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
242 ff_rtmp_packet_destroy(&pkt);
243 }
244
245
246 /**
247 * Generate 'deleteStream' call and send it to the server. It should make
248 * the server remove some channel for media streams.
249 */
250 static void gen_delete_stream(URLContext *s, RTMPContext *rt)
251 {
252 RTMPPacket pkt;
253 uint8_t *p;
254
255 av_log(s, AV_LOG_DEBUG, "Deleting stream...\n");
256 ff_rtmp_packet_create(&pkt, RTMP_SYSTEM_CHANNEL, RTMP_PT_INVOKE, 0, 34);
257
258 p = pkt.data;
259 ff_amf_write_string(&p, "deleteStream");
260 ff_amf_write_number(&p, 0.0);
261 ff_amf_write_null(&p);
262 ff_amf_write_number(&p, rt->main_channel_id);
263
264 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
265 ff_rtmp_packet_destroy(&pkt);
266 }
267
268 /**
269 * Generate 'play' call and send it to the server, then ping the server
270 * to start actual playing.
271 */
272 static void gen_play(URLContext *s, RTMPContext *rt)
273 {
274 RTMPPacket pkt;
275 uint8_t *p;
276
277 av_log(s, AV_LOG_DEBUG, "Sending play command for '%s'\n", rt->playpath);
278 ff_rtmp_packet_create(&pkt, RTMP_VIDEO_CHANNEL, RTMP_PT_INVOKE, 0,
279 20 + strlen(rt->playpath));
280 pkt.extra = rt->main_channel_id;
281
282 p = pkt.data;
283 ff_amf_write_string(&p, "play");
284 ff_amf_write_number(&p, 0.0);
285 ff_amf_write_null(&p);
286 ff_amf_write_string(&p, rt->playpath);
287
288 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
289 ff_rtmp_packet_destroy(&pkt);
290
291 // set client buffer time disguised in ping packet
292 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, 1, 10);
293
294 p = pkt.data;
295 bytestream_put_be16(&p, 3);
296 bytestream_put_be32(&p, 1);
297 bytestream_put_be32(&p, 256); //TODO: what is a good value here?
298
299 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
300 ff_rtmp_packet_destroy(&pkt);
301 }
302
303 /**
304 * Generate 'publish' call and send it to the server.
305 */
306 static void gen_publish(URLContext *s, RTMPContext *rt)
307 {
308 RTMPPacket pkt;
309 uint8_t *p;
310
311 av_log(s, AV_LOG_DEBUG, "Sending publish command for '%s'\n", rt->playpath);
312 ff_rtmp_packet_create(&pkt, RTMP_SOURCE_CHANNEL, RTMP_PT_INVOKE, 0,
313 30 + strlen(rt->playpath));
314 pkt.extra = rt->main_channel_id;
315
316 p = pkt.data;
317 ff_amf_write_string(&p, "publish");
318 ff_amf_write_number(&p, 0.0);
319 ff_amf_write_null(&p);
320 ff_amf_write_string(&p, rt->playpath);
321 ff_amf_write_string(&p, "live");
322
323 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
324 ff_rtmp_packet_destroy(&pkt);
325 }
326
327 /**
328 * Generate ping reply and send it to the server.
329 */
330 static void gen_pong(URLContext *s, RTMPContext *rt, RTMPPacket *ppkt)
331 {
332 RTMPPacket pkt;
333 uint8_t *p;
334
335 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_PING, ppkt->timestamp + 1, 6);
336 p = pkt.data;
337 bytestream_put_be16(&p, 7);
338 bytestream_put_be32(&p, AV_RB32(ppkt->data+2));
339 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
340 ff_rtmp_packet_destroy(&pkt);
341 }
342
343 /**
344 * Generate report on bytes read so far and send it to the server.
345 */
346 static void gen_bytes_read(URLContext *s, RTMPContext *rt, uint32_t ts)
347 {
348 RTMPPacket pkt;
349 uint8_t *p;
350
351 ff_rtmp_packet_create(&pkt, RTMP_NETWORK_CHANNEL, RTMP_PT_BYTES_READ, ts, 4);
352 p = pkt.data;
353 bytestream_put_be32(&p, rt->bytes_read);
354 ff_rtmp_packet_write(rt->stream, &pkt, rt->chunk_size, rt->prev_pkt[1]);
355 ff_rtmp_packet_destroy(&pkt);
356 }
357
358 //TODO: Move HMAC code somewhere. Eventually.
359 #define HMAC_IPAD_VAL 0x36
360 #define HMAC_OPAD_VAL 0x5C
361
362 /**
363 * Calculate HMAC-SHA2 digest for RTMP handshake packets.
364 *
365 * @param src input buffer
366 * @param len input buffer length (should be 1536)
367 * @param gap offset in buffer where 32 bytes should not be taken into account
368 * when calculating digest (since it will be used to store that digest)
369 * @param key digest key
370 * @param keylen digest key length
371 * @param dst buffer where calculated digest will be stored (32 bytes)
372 */
373 static void rtmp_calc_digest(const uint8_t *src, int len, int gap,
374 const uint8_t *key, int keylen, uint8_t *dst)
375 {
376 struct AVSHA *sha;
377 uint8_t hmac_buf[64+32] = {0};
378 int i;
379
380 sha = av_mallocz(av_sha_size);
381
382 if (keylen < 64) {
383 memcpy(hmac_buf, key, keylen);
384 } else {
385 av_sha_init(sha, 256);
386 av_sha_update(sha,key, keylen);
387 av_sha_final(sha, hmac_buf);
388 }
389 for (i = 0; i < 64; i++)
390 hmac_buf[i] ^= HMAC_IPAD_VAL;
391
392 av_sha_init(sha, 256);
393 av_sha_update(sha, hmac_buf, 64);
394 if (gap <= 0) {
395 av_sha_update(sha, src, len);
396 } else { //skip 32 bytes used for storing digest
397 av_sha_update(sha, src, gap);
398 av_sha_update(sha, src + gap + 32, len - gap - 32);
399 }
400 av_sha_final(sha, hmac_buf + 64);
401
402 for (i = 0; i < 64; i++)
403 hmac_buf[i] ^= HMAC_IPAD_VAL ^ HMAC_OPAD_VAL; //reuse XORed key for opad
404 av_sha_init(sha, 256);
405 av_sha_update(sha, hmac_buf, 64+32);
406 av_sha_final(sha, dst);
407
408 av_free(sha);
409 }
410
411 /**
412 * Put HMAC-SHA2 digest of packet data (except for the bytes where this digest
413 * will be stored) into that packet.
414 *
415 * @param buf handshake data (1536 bytes)
416 * @return offset to the digest inside input data
417 */
418 static int rtmp_handshake_imprint_with_digest(uint8_t *buf)
419 {
420 int i, digest_pos = 0;
421
422 for (i = 8; i < 12; i++)
423 digest_pos += buf[i];
424 digest_pos = (digest_pos % 728) + 12;
425
426 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
427 rtmp_player_key, PLAYER_KEY_OPEN_PART_LEN,
428 buf + digest_pos);
429 return digest_pos;
430 }
431
432 /**
433 * Verify that the received server response has the expected digest value.
434 *
435 * @param buf handshake data received from the server (1536 bytes)
436 * @param off position to search digest offset from
437 * @return 0 if digest is valid, digest position otherwise
438 */
439 static int rtmp_validate_digest(uint8_t *buf, int off)
440 {
441 int i, digest_pos = 0;
442 uint8_t digest[32];
443
444 for (i = 0; i < 4; i++)
445 digest_pos += buf[i + off];
446 digest_pos = (digest_pos % 728) + off + 4;
447
448 rtmp_calc_digest(buf, RTMP_HANDSHAKE_PACKET_SIZE, digest_pos,
449 rtmp_server_key, SERVER_KEY_OPEN_PART_LEN,
450 digest);
451 if (!memcmp(digest, buf + digest_pos, 32))
452 return digest_pos;
453 return 0;
454 }
455
456 /**
457 * Perform handshake with the server by means of exchanging pseudorandom data
458 * signed with HMAC-SHA2 digest.
459 *
460 * @return 0 if handshake succeeds, negative value otherwise
461 */
462 static int rtmp_handshake(URLContext *s, RTMPContext *rt)
463 {
464 AVLFG rnd;
465 uint8_t tosend [RTMP_HANDSHAKE_PACKET_SIZE+1] = {
466 3, // unencrypted data
467 0, 0, 0, 0, // client uptime
468 RTMP_CLIENT_VER1,
469 RTMP_CLIENT_VER2,
470 RTMP_CLIENT_VER3,
471 RTMP_CLIENT_VER4,
472 };
473 uint8_t clientdata[RTMP_HANDSHAKE_PACKET_SIZE];
474 uint8_t serverdata[RTMP_HANDSHAKE_PACKET_SIZE+1];
475 int i;
476 int server_pos, client_pos;
477 uint8_t digest[32];
478
479 av_log(s, AV_LOG_DEBUG, "Handshaking...\n");
480
481 av_lfg_init(&rnd, 0xDEADC0DE);
482 // generate handshake packet - 1536 bytes of pseudorandom data
483 for (i = 9; i <= RTMP_HANDSHAKE_PACKET_SIZE; i++)
484 tosend[i] = av_lfg_get(&rnd) >> 24;
485 client_pos = rtmp_handshake_imprint_with_digest(tosend + 1);
486
487 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE + 1);
488 i = ffurl_read_complete(rt->stream, serverdata, RTMP_HANDSHAKE_PACKET_SIZE + 1);
489 if (i != RTMP_HANDSHAKE_PACKET_SIZE + 1) {
490 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
491 return -1;
492 }
493 i = ffurl_read_complete(rt->stream, clientdata, RTMP_HANDSHAKE_PACKET_SIZE);
494 if (i != RTMP_HANDSHAKE_PACKET_SIZE) {
495 av_log(s, AV_LOG_ERROR, "Cannot read RTMP handshake response\n");
496 return -1;
497 }
498
499 av_log(s, AV_LOG_DEBUG, "Server version %d.%d.%d.%d\n",
500 serverdata[5], serverdata[6], serverdata[7], serverdata[8]);
501
502 if (rt->is_input && serverdata[5] >= 3) {
503 server_pos = rtmp_validate_digest(serverdata + 1, 772);
504 if (!server_pos) {
505 server_pos = rtmp_validate_digest(serverdata + 1, 8);
506 if (!server_pos) {
507 av_log(s, AV_LOG_ERROR, "Server response validating failed\n");
508 return -1;
509 }
510 }
511
512 rtmp_calc_digest(tosend + 1 + client_pos, 32, 0,
513 rtmp_server_key, sizeof(rtmp_server_key),
514 digest);
515 rtmp_calc_digest(clientdata, RTMP_HANDSHAKE_PACKET_SIZE-32, 0,
516 digest, 32,
517 digest);
518 if (memcmp(digest, clientdata + RTMP_HANDSHAKE_PACKET_SIZE - 32, 32)) {
519 av_log(s, AV_LOG_ERROR, "Signature mismatch\n");
520 return -1;
521 }
522
523 for (i = 0; i < RTMP_HANDSHAKE_PACKET_SIZE; i++)
524 tosend[i] = av_lfg_get(&rnd) >> 24;
525 rtmp_calc_digest(serverdata + 1 + server_pos, 32, 0,
526 rtmp_player_key, sizeof(rtmp_player_key),
527 digest);
528 rtmp_calc_digest(tosend, RTMP_HANDSHAKE_PACKET_SIZE - 32, 0,
529 digest, 32,
530 tosend + RTMP_HANDSHAKE_PACKET_SIZE - 32);
531
532 // write reply back to the server
533 ffurl_write(rt->stream, tosend, RTMP_HANDSHAKE_PACKET_SIZE);
534 } else {
535 ffurl_write(rt->stream, serverdata+1, RTMP_HANDSHAKE_PACKET_SIZE);
536 }
537
538 return 0;
539 }
540
541 /**
542 * Parse received packet and possibly perform some action depending on
543 * the packet contents.
544 * @return 0 for no errors, negative values for serious errors which prevent
545 * further communications, positive values for uncritical errors
546 */
547 static int rtmp_parse_result(URLContext *s, RTMPContext *rt, RTMPPacket *pkt)
548 {
549 int i, t;
550 const uint8_t *data_end = pkt->data + pkt->data_size;
551
552 #ifdef DEBUG
553 ff_rtmp_packet_dump(s, pkt);
554 #endif
555
556 switch (pkt->type) {
557 case RTMP_PT_CHUNK_SIZE:
558 if (pkt->data_size != 4) {
559 av_log(s, AV_LOG_ERROR,
560 "Chunk size change packet is not 4 bytes long (%d)\n", pkt->data_size);
561 return -1;
562 }
563 if (!rt->is_input)
564 ff_rtmp_packet_write(rt->stream, pkt, rt->chunk_size, rt->prev_pkt[1]);
565 rt->chunk_size = AV_RB32(pkt->data);
566 if (rt->chunk_size <= 0) {
567 av_log(s, AV_LOG_ERROR, "Incorrect chunk size %d\n", rt->chunk_size);
568 return -1;
569 }
570 av_log(s, AV_LOG_DEBUG, "New chunk size = %d\n", rt->chunk_size);
571 break;
572 case RTMP_PT_PING:
573 t = AV_RB16(pkt->data);
574 if (t == 6)
575 gen_pong(s, rt, pkt);
576 break;
577 case RTMP_PT_CLIENT_BW:
578 if (pkt->data_size < 4) {
579 av_log(s, AV_LOG_ERROR,
580 "Client bandwidth report packet is less than 4 bytes long (%d)\n",
581 pkt->data_size);
582 return -1;
583 }
584 av_log(s, AV_LOG_DEBUG, "Client bandwidth = %d\n", AV_RB32(pkt->data));
585 rt->client_report_size = AV_RB32(pkt->data) >> 1;
586 break;
587 case RTMP_PT_INVOKE:
588 //TODO: check for the messages sent for wrong state?
589 if (!memcmp(pkt->data, "\002\000\006_error", 9)) {
590 uint8_t tmpstr[256];
591
592 if (!ff_amf_get_field_value(pkt->data + 9, data_end,
593 "description", tmpstr, sizeof(tmpstr)))
594 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
595 return -1;
596 } else if (!memcmp(pkt->data, "\002\000\007_result", 10)) {
597 switch (rt->state) {
598 case STATE_HANDSHAKED:
599 if (!rt->is_input) {
600 gen_release_stream(s, rt);
601 gen_fcpublish_stream(s, rt);
602 rt->state = STATE_RELEASING;
603 } else {
604 rt->state = STATE_CONNECTING;
605 }
606 gen_create_stream(s, rt);
607 break;
608 case STATE_FCPUBLISH:
609 rt->state = STATE_CONNECTING;
610 break;
611 case STATE_RELEASING:
612 rt->state = STATE_FCPUBLISH;
613 /* hack for Wowza Media Server, it does not send result for
614 * releaseStream and FCPublish calls */
615 if (!pkt->data[10]) {
616 int pkt_id = (int) av_int2dbl(AV_RB64(pkt->data + 11));
617 if (pkt_id == 4)
618 rt->state = STATE_CONNECTING;
619 }
620 if (rt->state != STATE_CONNECTING)
621 break;
622 case STATE_CONNECTING:
623 //extract a number from the result
624 if (pkt->data[10] || pkt->data[19] != 5 || pkt->data[20]) {
625 av_log(s, AV_LOG_WARNING, "Unexpected reply on connect()\n");
626 } else {
627 rt->main_channel_id = (int) av_int2dbl(AV_RB64(pkt->data + 21));
628 }
629 if (rt->is_input) {
630 gen_play(s, rt);
631 } else {
632 gen_publish(s, rt);
633 }
634 rt->state = STATE_READY;
635 break;
636 }
637 } else if (!memcmp(pkt->data, "\002\000\010onStatus", 11)) {
638 const uint8_t* ptr = pkt->data + 11;
639 uint8_t tmpstr[256];
640
641 for (i = 0; i < 2; i++) {
642 t = ff_amf_tag_size(ptr, data_end);
643 if (t < 0)
644 return 1;
645 ptr += t;
646 }
647 t = ff_amf_get_field_value(ptr, data_end,
648 "level", tmpstr, sizeof(tmpstr));
649 if (!t && !strcmp(tmpstr, "error")) {
650 if (!ff_amf_get_field_value(ptr, data_end,
651 "description", tmpstr, sizeof(tmpstr)))
652 av_log(s, AV_LOG_ERROR, "Server error: %s\n",tmpstr);
653 return -1;
654 }
655 t = ff_amf_get_field_value(ptr, data_end,
656 "code", tmpstr, sizeof(tmpstr));
657 if (!t && !strcmp(tmpstr, "NetStream.Play.Start")) rt->state = STATE_PLAYING;
658 if (!t && !strcmp(tmpstr, "NetStream.Play.Stop")) rt->state = STATE_STOPPED;
659 if (!t && !strcmp(tmpstr, "NetStream.Play.UnpublishNotify")) rt->state = STATE_STOPPED;
660 if (!t && !strcmp(tmpstr, "NetStream.Publish.Start")) rt->state = STATE_PUBLISHING;
661 }
662 break;
663 }
664 return 0;
665 }
666
667 /**
668 * Interact with the server by receiving and sending RTMP packets until
669 * there is some significant data (media data or expected status notification).
670 *
671 * @param s reading context
672 * @param for_header non-zero value tells function to work until it
673 * gets notification from the server that playing has been started,
674 * otherwise function will work until some media data is received (or
675 * an error happens)
676 * @return 0 for successful operation, negative value in case of error
677 */
678 static int get_packet(URLContext *s, int for_header)
679 {
680 RTMPContext *rt = s->priv_data;
681 int ret;
682 uint8_t *p;
683 const uint8_t *next;
684 uint32_t data_size;
685 uint32_t ts, cts, pts=0;
686
687 if (rt->state == STATE_STOPPED)
688 return AVERROR_EOF;
689
690 for (;;) {
691 RTMPPacket rpkt = { 0 };
692 if ((ret = ff_rtmp_packet_read(rt->stream, &rpkt,
693 rt->chunk_size, rt->prev_pkt[0])) <= 0) {
694 if (ret == 0) {
695 return AVERROR(EAGAIN);
696 } else {
697 return AVERROR(EIO);
698 }
699 }
700 rt->bytes_read += ret;
701 if (rt->bytes_read > rt->last_bytes_read + rt->client_report_size) {
702 av_log(s, AV_LOG_DEBUG, "Sending bytes read report\n");
703 gen_bytes_read(s, rt, rpkt.timestamp + 1);
704 rt->last_bytes_read = rt->bytes_read;
705 }
706
707 ret = rtmp_parse_result(s, rt, &rpkt);
708 if (ret < 0) {//serious error in current packet
709 ff_rtmp_packet_destroy(&rpkt);
710 return -1;
711 }
712 if (rt->state == STATE_STOPPED) {
713 ff_rtmp_packet_destroy(&rpkt);
714 return AVERROR_EOF;
715 }
716 if (for_header && (rt->state == STATE_PLAYING || rt->state == STATE_PUBLISHING)) {
717 ff_rtmp_packet_destroy(&rpkt);
718 return 0;
719 }
720 if (!rpkt.data_size || !rt->is_input) {
721 ff_rtmp_packet_destroy(&rpkt);
722 continue;
723 }
724 if (rpkt.type == RTMP_PT_VIDEO || rpkt.type == RTMP_PT_AUDIO ||
725 (rpkt.type == RTMP_PT_NOTIFY && !memcmp("\002\000\012onMetaData", rpkt.data, 13))) {
726 ts = rpkt.timestamp;
727
728 // generate packet header and put data into buffer for FLV demuxer
729 rt->flv_off = 0;
730 rt->flv_size = rpkt.data_size + 15;
731 rt->flv_data = p = av_realloc(rt->flv_data, rt->flv_size);
732 bytestream_put_byte(&p, rpkt.type);
733 bytestream_put_be24(&p, rpkt.data_size);
734 bytestream_put_be24(&p, ts);
735 bytestream_put_byte(&p, ts >> 24);
736 bytestream_put_be24(&p, 0);
737 bytestream_put_buffer(&p, rpkt.data, rpkt.data_size);
738 bytestream_put_be32(&p, 0);
739 ff_rtmp_packet_destroy(&rpkt);
740 return 0;
741 } else if (rpkt.type == RTMP_PT_METADATA) {
742 // we got raw FLV data, make it available for FLV demuxer
743 rt->flv_off = 0;
744 rt->flv_size = rpkt.data_size;
745 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
746 /* rewrite timestamps */
747 next = rpkt.data;
748 ts = rpkt.timestamp;
749 while (next - rpkt.data < rpkt.data_size - 11) {
750 next++;
751 data_size = bytestream_get_be24(&next);
752 p=next;
753 cts = bytestream_get_be24(&next);
754 cts |= bytestream_get_byte(&next) << 24;
755 if (pts==0)
756 pts=cts;
757 ts += cts - pts;
758 pts = cts;
759 bytestream_put_be24(&p, ts);
760 bytestream_put_byte(&p, ts >> 24);
761 next += data_size + 3 + 4;
762 }
763 memcpy(rt->flv_data, rpkt.data, rpkt.data_size);
764 ff_rtmp_packet_destroy(&rpkt);
765 return 0;
766 }
767 ff_rtmp_packet_destroy(&rpkt);
768 }
769 }
770
771 static int rtmp_close(URLContext *h)
772 {
773 RTMPContext *rt = h->priv_data;
774
775 if (!rt->is_input) {
776 rt->flv_data = NULL;
777 if (rt->out_pkt.data_size)
778 ff_rtmp_packet_destroy(&rt->out_pkt);
779 if (rt->state > STATE_FCPUBLISH)
780 gen_fcunpublish_stream(h, rt);
781 }
782 if (rt->state > STATE_HANDSHAKED)
783 gen_delete_stream(h, rt);
784
785 av_freep(&rt->flv_data);
786 ffurl_close(rt->stream);
787 av_free(rt);
788 return 0;
789 }
790
791 /**
792 * Open RTMP connection and verify that the stream can be played.
793 *
794 * URL syntax: rtmp://server[:port][/app][/playpath]
795 * where 'app' is first one or two directories in the path
796 * (e.g. /ondemand/, /flash/live/, etc.)
797 * and 'playpath' is a file name (the rest of the path,
798 * may be prefixed with "mp4:")
799 */
800 static int rtmp_open(URLContext *s, const char *uri, int flags)
801 {
802 RTMPContext *rt;
803 char proto[8], hostname[256], path[1024], *fname;
804 uint8_t buf[2048];
805 int port;
806 int ret;
807
808 rt = av_mallocz(sizeof(RTMPContext));
809 if (!rt)
810 return AVERROR(ENOMEM);
811 s->priv_data = rt;
812 rt->is_input = !(flags & AVIO_FLAG_WRITE);
813
814 av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), &port,
815 path, sizeof(path), s->filename);
816
817 if (port < 0)
818 port = RTMP_DEFAULT_PORT;
819 ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
820
821 if (ffurl_open(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
822 &s->interrupt_callback, NULL) < 0) {
823 av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
824 goto fail;
825 }
826
827 rt->state = STATE_START;
828 if (rtmp_handshake(s, rt))
829 return -1;
830
831 rt->chunk_size = 128;
832 rt->state = STATE_HANDSHAKED;
833 //extract "app" part from path
834 if (!strncmp(path, "/ondemand/", 10)) {
835 fname = path + 10;
836 memcpy(rt->app, "ondemand", 9);
837 } else {
838 char *p = strchr(path + 1, '/');
839 if (!p) {
840 fname = path + 1;
841 rt->app[0] = '\0';
842 } else {
843 char *c = strchr(p + 1, ':');
844 fname = strchr(p + 1, '/');
845 if (!fname || c < fname) {
846 fname = p + 1;
847 av_strlcpy(rt->app, path + 1, p - path);
848 } else {
849 fname++;
850 av_strlcpy(rt->app, path + 1, fname - path - 1);
851 }
852 }
853 }
854 if (!strchr(fname, ':') &&
855 (!strcmp(fname + strlen(fname) - 4, ".f4v") ||
856 !strcmp(fname + strlen(fname) - 4, ".mp4"))) {
857 memcpy(rt->playpath, "mp4:", 5);
858 } else {
859 rt->playpath[0] = 0;
860 }
861 strncat(rt->playpath, fname, sizeof(rt->playpath) - 5);
862
863 rt->client_report_size = 1048576;
864 rt->bytes_read = 0;
865 rt->last_bytes_read = 0;
866
867 av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
868 proto, path, rt->app, rt->playpath);
869 gen_connect(s, rt, proto, hostname, port);
870
871 do {
872 ret = get_packet(s, 1);
873 } while (ret == EAGAIN);
874 if (ret < 0)
875 goto fail;
876
877 if (rt->is_input) {
878 // generate FLV header for demuxer
879 rt->flv_size = 13;
880 rt->flv_data = av_realloc(rt->flv_data, rt->flv_size);
881 rt->flv_off = 0;
882 memcpy(rt->flv_data, "FLV\1\5\0\0\0\011\0\0\0\0", rt->flv_size);
883 } else {
884 rt->flv_size = 0;
885 rt->flv_data = NULL;
886 rt->flv_off = 0;
887 rt->skip_bytes = 13;
888 }
889
890 s->max_packet_size = rt->stream->max_packet_size;
891 s->is_streamed = 1;
892 return 0;
893
894 fail:
895 rtmp_close(s);
896 return AVERROR(EIO);
897 }
898
899 static int rtmp_read(URLContext *s, uint8_t *buf, int size)
900 {
901 RTMPContext *rt = s->priv_data;
902 int orig_size = size;
903 int ret;
904
905 while (size > 0) {
906 int data_left = rt->flv_size - rt->flv_off;
907
908 if (data_left >= size) {
909 memcpy(buf, rt->flv_data + rt->flv_off, size);
910 rt->flv_off += size;
911 return orig_size;
912 }
913 if (data_left > 0) {
914 memcpy(buf, rt->flv_data + rt->flv_off, data_left);
915 buf += data_left;
916 size -= data_left;
917 rt->flv_off = rt->flv_size;
918 return data_left;
919 }
920 if ((ret = get_packet(s, 0)) < 0)
921 return ret;
922 }
923 return orig_size;
924 }
925
926 static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
927 {
928 RTMPContext *rt = s->priv_data;
929 int size_temp = size;
930 int pktsize, pkttype;
931 uint32_t ts;
932 const uint8_t *buf_temp = buf;
933
934 do {
935 if (rt->skip_bytes) {
936 int skip = FFMIN(rt->skip_bytes, size_temp);
937 buf_temp += skip;
938 size_temp -= skip;
939 rt->skip_bytes -= skip;
940 continue;
941 }
942
943 if (rt->flv_header_bytes < 11) {
944 const uint8_t *header = rt->flv_header;
945 int copy = FFMIN(11 - rt->flv_header_bytes, size_temp);
946 bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
947 rt->flv_header_bytes += copy;
948 size_temp -= copy;
949 if (rt->flv_header_bytes < 11)
950 break;
951
952 pkttype = bytestream_get_byte(&header);
953 pktsize = bytestream_get_be24(&header);
954 ts = bytestream_get_be24(&header);
955 ts |= bytestream_get_byte(&header) << 24;
956 bytestream_get_be24(&header);
957 rt->flv_size = pktsize;
958
959 //force 12bytes header
960 if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
961 pkttype == RTMP_PT_NOTIFY) {
962 if (pkttype == RTMP_PT_NOTIFY)
963 pktsize += 16;
964 rt->prev_pkt[1][RTMP_SOURCE_CHANNEL].channel_id = 0;
965 }
966
967 //this can be a big packet, it's better to send it right here
968 ff_rtmp_packet_create(&rt->out_pkt, RTMP_SOURCE_CHANNEL, pkttype, ts, pktsize);
969 rt->out_pkt.extra = rt->main_channel_id;
970 rt->flv_data = rt->out_pkt.data;
971
972 if (pkttype == RTMP_PT_NOTIFY)
973 ff_amf_write_string(&rt->flv_data, "@setDataFrame");
974 }
975
976 if (rt->flv_size - rt->flv_off > size_temp) {
977 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, size_temp);
978 rt->flv_off += size_temp;
979 size_temp = 0;
980 } else {
981 bytestream_get_buffer(&buf_temp, rt->flv_data + rt->flv_off, rt->flv_size - rt->flv_off);
982 size_temp -= rt->flv_size - rt->flv_off;
983 rt->flv_off += rt->flv_size - rt->flv_off;
984 }
985
986 if (rt->flv_off == rt->flv_size) {
987 rt->skip_bytes = 4;
988
989 ff_rtmp_packet_write(rt->stream, &rt->out_pkt, rt->chunk_size, rt->prev_pkt[1]);
990 ff_rtmp_packet_destroy(&rt->out_pkt);
991 rt->flv_size = 0;
992 rt->flv_off = 0;
993 rt->flv_header_bytes = 0;
994 }
995 } while (buf_temp - buf < size);
996 return size;
997 }
998
999 URLProtocol ff_rtmp_protocol = {
1000 .name = "rtmp",
1001 .url_open = rtmp_open,
1002 .url_read = rtmp_read,
1003 .url_write = rtmp_write,
1004 .url_close = rtmp_close,
1005 };